1. Field of the Invention
This invention relates generally to telecommunication, and more particularly to high density telecommunication testing.
2. Description of the Related Art
Today, modern telecommunications systems often perform complex operations, such as data compression, when transmitting signals through the telecommunications network. These operations generally have non-linear effects on the signal inputs. As a result, it is often not possible to model the effects of the network by simulating the additive affect of each component of the network. In particular, the affect of the network on speech is not easily derivable from studying the network's affect on a simple test signal such as a sine wave.
Hence, voice communication signals generally are tested using voice generation and analyzing equipment in the form of a telecommunication testing system. FIG. 1 is a block diagram showing an exemplary conventional telecommunication testing configuration 100. As shown in FIG. 1, the telecommunication testing configuration 100 includes a system under test (SUT) 102, such as a telecommunication system, in communication with a test system 104. As mentioned above, one technique for testing the SUT 102 for voice QoS is call generation.
Call Generation is a testing mode in which the test system 104 creates telephone traffic by executing compiled call sequences (scripts). To test the SUT 102, the test system 104 provides a maximal load on the SUT 102. In particular, the test system 104 places data on the input channels of the SUT 102, and receives and analyzes the output data quality of the SUT 102 in real time.
In order to reduce the amount of data passing through the communication lines, the data is compressed before transmitting and decompressed after receiving using speech codecs, often referred to as vocoders. As shown in FIG. 1, the SUT 102 includes a codec 106a, and the test system 104 includes a similar codec 106b. In this manner, the test system 104 can encode speech data using the codec 106b. The test system 104 then transmits the encoded speech data to the SUT 102, which decodes the speech data using the codec 106a of the SUT 102. Similarly, the SUT 102 encodes speech data using the codec 106a and transmits the encoded speech data to the test system 104. The test system then decodes the speech data using the codec 106b of the test system 104.
FIG. 2 shows an exemplary conventional speech codec 106 for encoding and decoding speech data. The speech codec 200 is a hardware circuit (chip) or software/firmware routine that converts the spoken word into digital code and vice versa. In particular, a speech codec is an audio codec specialized for human voice. By analyzing vocal tract sounds, a recipe for rebuilding the sound at the other end is sent rather than the soundwaves themselves. As a result, the speech codec is able to achieve a much higher compression ratio than regular audio codecs, which yields a smaller amount of digital data for transmission.
As shown in FIG. 2, the speech codec 106 includes an encoder 200 and a decoder 202. The codec 106 both encodes and decodes speech data using the encoder 200 and the decoder 202 respectively. For example, in a SUT, the codec 106 can be used to transform data between Pulse Code Modulation (PCM) format and Adaptive Differential PCM (ADPCM) format.
PCM is a technique for converting analog signals into digital form that is widely used by the telephone companies in their T1 circuits. For example, telephone conversations, as well as data transmissions via modem, are converted into digital via PCM for transport over high-speed intercity trunks. In North America and Japan, PCM samples the analog waves 8,000 times per second and converts each sample into an 8-bit number, resulting in a 64 Kbps data stream (a single DS0 channel). The sampling rate is twice the 4 kHz bandwidth required for a toll-quality conversation. ADPCM is an advanced PCM technique that converts analog sound into digital data and vice versa. Instead of coding an absolute measurement at each sample point, it codes the difference between samples and can dynamically switch the coding scale to compensate for variations in amplitude and frequency.
Thus, for example, the decoder 202 section of the codec 106 can receive a PCM signal from the telecommunications network. Once received, the decoder 202 can decode the PCM signal and provide the uncompressed speech data to the telecommunications system, which processes the signal. Thereafter, the telecommunications system uses the encoder 200 to encode the uncompressed data into, for example, an ADPCM signal and transmits. In this manner, the codec 106 allows a system to receive and process PCM data and transmit ADPCM data. To test such a system, the test system can include an encoder that encodes PCM data and a decoder that decodes ADPCM data.
For example, referring back to FIG. 1, when testing the SUT 102, the test system 104 encodes speech data using the codec 106b. For example, the codec can encode uncompressed speech test data into PCM format. The test system then transmits the encoded PCM data to the SUT 102, which uses the codec 106a to decode the PCM data for processing. The SUT 102 can then encode the speech data into, for example, ADPCM format and transmit the encoded data back to the test system 104. The test system then decodes the ADPCM speech data using the codec 106b and analyses the speech data for quality.
Unfortunately, test systems 104 using call generation typically cannot support a large amount of data channels without distorting the performance of the SUT 102. For example, if the SUT 102 can support, for example, 300 simultaneous data channels, a typical testing system 104 can only support, for example, about 100 simultaneous data channels. As a result, three testing systems 104 would be needed to test the performance of the SUT 102.
In view of the foregoing, there is a need for systems and methods for high density telecommunication testing. The systems and methods should be capable of performing quality of service (QoS) testing on the SUT, and further, should support an increased number of simultaneous data channels without distorting the performance of the SUT.